sip协议英文资料

Network Working Group                                       J. RosenbergRequest for Comments: 3261                                   dynamicsoftObsoletes: 2543                                           H. SchulzrinneCategory: Standards Track                                    Columbia U.                                                            G. Camarillo                                                                Ericsson                                                             A. Johnston                                                                WorldCom                                                             J. Peterson                                                                 Neustar                                                               R. Sparks                                                             dynamicsoft                                                              M. Handley                                                                    ICIR                                                             E. Schooler                                                                    AT&T                                                               June 2002                    SIP: Session Initiation ProtocolStatus of this Memo   This document specifies an Internet standards track protocol for the   Internet community, and requests discussion and suggestions for   improvements.  Please refer to the current edition of the "Internet   Official Protocol Standards" (STD 1) for the standardization state   and status of this protocol.  Distribution of this memo is unlimited.Copyright Notice   Copyright (C) The Internet Society (2002).  All Rights Reserved.Abstract   This document describes Session Initiation Protocol (SIP), an   application-layer control (signaling) protocol for creating,   modifying, and terminating sessions with one or more participants.   These sessions include Internet telephone calls, multimedia   distribution, and multimedia conferences.   SIP invitations used to create sessions carry session descriptions   that allow participants to agree on a set of compatible media types.   SIP makes use of elements called proxy servers to help route requests   to the user's current location, authenticate and authorize users for   services, implement provider call-routing policies, and provide   features to users.  SIP also provides a registration function that   allows users to upload their current locations for use by proxy   servers.  SIP runs on top of several different transport protocols.Rosenberg, et. al.          Standards Track                     [Page 1]RFC 3261            SIP: Session Initiation Protocol           June 2002Table of Contents   1          Introduction ........................................    8   2          Overview of SIP Functionality .......................    9   3          Terminology .........................................   10   4          Overview of Operation ...............................   10   5          Structure of the Protocol ...........................   18   6          Definitions .........................................   20   7          SIP Messages ........................................   26   7.1        Requests ............................................   27   7.2        Responses ...........................................   28   7.3        Header Fields .......................................   29   7.3.1      Header Field Format .................................   30   7.3.2      Header Field Classification .........................   32   7.3.3      Compact Form ........................................   32   7.4        Bodies ..............................................   33   7.4.1      Message Body Type ...................................   33   7.4.2      Message Body Length .................................   33   7.5        Framing SIP Messages ................................   34   8          General User Agent Behavior .........................   34   8.1        UAC Behavior ........................................   35   8.1.1      Generating the Request ..............................   35   8.1.1.1    Request-URI .........................................   35   8.1.1.2    To ..................................................   36   8.1.1.3    From ................................................   37   8.1.1.4    Call-ID .............................................   37   8.1.1.5    CSeq ................................................   38   8.1.1.6    Max-Forwards ........................................   38   8.1.1.7    Via .................................................   39   8.1.1.8    Contact .............................................   40   8.1.1.9    Supported and Require ...............................   40   8.1.1.10   Additional Message Components .......................   41   8.1.2      Sending the Request .................................   41   8.1.3      Processing Responses ................................   42   8.1.3.1    Transaction Layer Errors ............................   42   8.1.3.2    Unrecognized Responses ..............................   42   8.1.3.3    Vias ................................................   43   8.1.3.4    Processing 3xx Responses ............................   43   8.1.3.5    Processing 4xx Responses ............................   45   8.2        UAS Behavior ........................................   46   8.2.1      Method Inspection ...................................   46   8.2.2      Header Inspection ...................................   46   8.2.2.1    To and Request-URI ..................................   46   8.2.2.2    Merged Requests .....................................   47   8.2.2.3    Require .............................................   47   8.2.3      Content Processing ..................................   48   8.2.4      Applying Extensions .................................   49   8.2.5      Processing the Request ..............................   49Rosenberg, et. al.          Standards Track                     [Page 2]RFC 3261            SIP: Session Initiation Protocol           June 2002   8.2.6      Generating the Response .............................   49   8.2.6.1    Sending a Provisional Response ......................   49   8.2.6.2    Headers and Tags ....................................   50   8.2.7      Stateless UAS Behavior ..............................   50   8.3        Redirect Servers ....................................   51   9          Canceling a Request .................................   53   9.1        Client Behavior .....................................   53   9.2        Server Behavior .....................................   55   10         Registrations .......................................   56   10.1       Overview ............................................   56   10.2       Constructing the REGISTER Request ...................   57   10.2.1     Adding Bindings .....................................   59   10.2.1.1   Setting the Expiration Interval of Contact Addresses    60   10.2.1.2   Preferences among Contact Addresses .................   61   10.2.2     Removing Bindings ...................................   61   10.2.3     Fetching Bindings ...................................   61   10.2.4     Refreshing Bindings .................................   61   10.2.5     Setting the Internal Clock ..........................   62   10.2.6     Discovering a Registrar .............................   62   10.2.7     Transmitting a Request ..............................   62   10.2.8     Error Responses .....................................   63   10.3       Processing REGISTER Requests ........................   63   11         Querying for Capabilities ...........................   66   11.1       Construction of OPTIONS Request .....................   67   11.2       Processing of OPTIONS Request .......................   68   12         Dialogs .............................................   69   12.1       Creation of a Dialog ................................   70   12.1.1     UAS behavior ........................................   70   12.1.2     UAC Behavior ........................................   71   12.2       Requests within a Dialog ............................   72   12.2.1     UAC Behavior ........................................   73   12.2.1.1   Generating the Request ..............................   73   12.2.1.2   Processing the Responses ............................   75   12.2.2     UAS Behavior ........................................   76   12.3       Termination of a Dialog .............................   77   13         Initiating a Session ................................   77   13.1       Overview ............................................   77   13.2       UAC Processing ......................................   78   13.2.1     Creating the Initial INVITE .........................   78   13.2.2     Processing INVITE Responses .........................   81   13.2.2.1   1xx Responses .......................................   81   13.2.2.2   3xx Responses .......................................   81   13.2.2.3   4xx, 5xx and 6xx Responses ..........................   81   13.2.2.4   2xx Responses .......................................   82   13.3       UAS Processing ......................................   83   13.3.1     Processing of the INVITE ............................   83   13.3.1.1   Progress ............................................   84   13.3.1.2   The INVITE is Redirected ............................   84Rosenberg, et. al.          Standards Track                     [Page 3]RFC 3261            SIP: Session Initiation Protocol           June 2002   13.3.1.3   The INVITE is Rejected ..............................   85   13.3.1.4   The INVITE is Accepted ..............................   85   14         Modifying an Existing Session .......................   86   14.1       UAC Behavior ........................................   86   14.2       UAS Behavior ........................................   88   15         Terminating a Session ...............................   89   15.1       Terminating a Session with a BYE Request ............   90   15.1.1     UAC Behavior ........................................   90   15.1.2     UAS Behavior ........................................   91   16         Proxy Behavior ......................................   91   16.1       Overview ............................................   91   16.2       Stateful Proxy ......................................   92   16.3       Request Validation ..................................   94   16.4       Route Information Preprocessing .....................   96   16.5       Determining Request Targets .........................   97   16.6       Request Forwarding ..................................   99   16.7       Response Processing .................................  107   16.8       Processing Timer C ..................................  114   16.9       Handling Transport Errors ...........................  115   16.10      CANCEL Processing ...................................  115   16.11      Stateless Proxy .....................................  116   16.12      Summary of Proxy Route Processing ...................  118   16.12.1    Examples ............................................  118   16.12.1.1  Basic SIP Trapezoid .................................  118   16.12.1.2  Traversing a Strict-Routing Proxy ...................  120   16.12.1.3  Rewriting Record-Route Header Field Values ..........  121   17         Transactions ........................................  122   17.1       Client Transaction ..................................  124   17.1.1     INVITE Client Transaction ...........................  125   17.1.1.1   Overview of INVITE Transaction ......................  125   17.1.1.2   Formal Description ..................................  125   17.1.1.3   Construction of the ACK Request .....................  129   17.1.2     Non-INVITE Client Transaction .......................  130   17.1.2.1   Overview of the non-INVITE Transaction ..............  130   17.1.2.2   Formal Description ..................................  131   17.1.3     Matching Responses to Client Transactions ...........  132   17.1.4     Handling Transport Errors ...........................  133   17.2       Server Transaction ..................................  134   17.2.1     INVITE Server Transaction ...........................  134   17.2.2     Non-INVITE Server Transaction .......................  137   17.2.3     Matching Requests to Server Transactions ............  138   17.2.4     Handling Transport Errors ...........................  141   18         Transport ...........................................  141   18.1       Clients .............................................  142   18.1.1     Sending Requests ....................................  142   18.1.2     Receiving Responses .................................  144   18.2       Servers .............................................  145   18.2.1     Receiving Requests ..................................  145Rosenberg, et. al.          Standards Track                     [Page 4]RFC 3261            SIP: Session Initiation Protocol           June 2002   18.2.2     Sending Responses ...................................  146   18.3       Framing .............................................  147   18.4       Error Handling ......................................  147   19         Common Message Components ...........................  147   19.1       SIP and SIPS Uniform Resource Indicators ............  148   19.1.1     SIP and SIPS URI Components .........................  148   19.1.2     Character Escaping Requirements .....................  152   19.1.3     Example SIP and SIPS URIs ...........................  153   19.1.4     URI Comparison ......................................  153   19.1.5     Forming Requests from a URI .........................  156   19.1.6     Relating SIP URIs and tel URLs ......................  157   19.2       Option Tags .........................................  158   19.3       Tags ................................................  159   20         Header Fields .......................................  159   20.1       Accept ..............................................  161   20.2       Accept-Encoding .....................................  163   20.3       Accept-Language .....................................  164   20.4       Alert-Info ..........................................  164   20.5       Allow ...............................................  165   20.6       Authentication-Info .................................  165   20.7       Authorization .......................................  165   20.8       Call-ID .............................................  166   20.9       Call-Info ...........................................  166   20.10      Contact .............................................  167   20.11      Content-Disposition .................................  168   20.12      Content-Encoding ....................................  169   20.13      Content-Language ....................................  169   20.14      Content-Length ......................................  169   20.15      Content-Type ........................................  170   20.16      CSeq ................................................  170   20.17      Date ................................................  170   20.18      Error-Info ..........................................  171   20.19      Expires .............................................  171   20.20      From ................................................  172   20.21      In-Reply-To .........................................  172   20.22      Max-Forwards ........................................  173   20.23      Min-Expires .........................................  173   20.24      MIME-Version ........................................  173   20.25      Organization ........................................  174   20.26      Priority ............................................  174   20.27      Proxy-Authenticate ..................................  174   20.28      Proxy-Authorization .................................  175   20.29      Proxy-Require .......................................  175   20.30      Record-Route ........................................  175   20.31      Reply-To ............................................  176   20.32      Require .............................................  176   20.33      Retry-After .........................................  176   20.34      Route ...............................................  177Rosenberg, et. al.          Standards Track                     [Page 5]RFC 3261            SIP: Session Initiation Protocol           June 2002   20.35      Server ..............................................  177   20.36      Subject .............................................  177   20.37      Supported ...........................................  178   20.38      Timestamp ...........................................  178   20.39      To ..................................................  178   20.40      Unsupported .........................................  179   20.41      User-Agent ..........................................  179   20.42      Via .................................................  179   20.43      Warning .............................................  180   20.44      WWW-Authenticate ....................................  182   21         Response Codes ......................................  182   21.1       Provisional 1xx .....................................  182   21.1.1     100 Trying ..........................................  183   21.1.2     180 Ringing .........................................  183   21.1.3     181 Call Is Being Forwarded .........................  183   21.1.4     182 Queued ..........................................  183   21.1.5     183 Session Progress ................................  183   21.2       Successful 2xx ......................................  183   21.2.1     200 OK ..............................................  183   21.3       Redirection 3xx .....................................  184   21.3.1     300 Multiple Choices ................................  184   21.3.2     301 Moved Permanently ...............................  184   21.3.3     302 Moved Temporarily ...............................  184   21.3.4     305 Use Proxy .......................................  185   21.3.5     380 Alternative Service .............................  185   21.4       Request Failure 4xx .................................  185   21.4.1     400 Bad Request .....................................  185   21.4.2     401 Unauthorized ....................................  185   21.4.3     402 Payment Required ................................  186   21.4.4     403 Forbidden .......................................  186   21.4.5     404 Not Found .......................................  186   21.4.6     405 Method Not Allowed ..............................  186   21.4.7     406 Not Acceptable ..................................  186   21.4.8     407 Proxy Authentication Required ...................  186   21.4.9     408 Request Timeout .................................  186   21.4.10    410 Gone ............................................  187   21.4.11    413 Request Entity Too Large ........................  187   21.4.12    414 Request-URI Too Long ............................  187   21.4.13    415 Unsupported Media Type ..........................  187   21.4.14    416 Unsupported URI Scheme ..........................  187   21.4.15    420 Bad Extension ...................................  187   21.4.16    421 Extension Required ..............................  188   21.4.17    423 Interval Too Brief ..............................  188   21.4.18    480 Temporarily Unavailable .........................  188   21.4.19    481 Call/Transaction Does Not Exist .................  188   21.4.20    482 Loop Detected ...................................  188   21.4.21    483 Too Many Hops ...................................  189   21.4.22    484 Address Incomplete ..............................  189Rosenberg, et. al.          Standards Track                     [Page 6]RFC 3261            SIP: Session Initiation Protocol           June 2002   21.4.23    485 Ambiguous .......................................  189   21.4.24    486 Busy Here .......................................  189   21.4.25    487 Request Terminated ..............................  190   21.4.26    488 Not Acceptable Here .............................  190   21.4.27    491 Request Pending .................................  190   21.4.28    493 Undecipherable ..................................  190   21.5       Server Failure 5xx ..................................  190   21.5.1     500 Server Internal Error ...........................  190   21.5.2     501 Not Implemented .................................  191   21.5.3     502 Bad Gateway .....................................  191   21.5.4     503 Service Unavailable .............................  191   21.5.5     504 Server Time-out .................................  191   21.5.6     505 Version Not Supported ...........................  192   21.5.7     513 Message Too Large ...............................  192   21.6       Global Failures 6xx .................................  192   21.6.1     600 Busy Everywhere .................................  192   21.6.2     603 Decline .........................................  192   21.6.3     604 Does Not Exist Anywhere .........................  192   21.6.4     606 Not Acceptable ..................................  192   22         Usage of HTTP Authentication ........................  193   22.1       Framework ...........................................  193   22.2       User-to-User Authentication .........................  195   22.3       Proxy-to-User Authentication ........................  197   22.4       The Digest Authentication Scheme ....................  199   23         S/MIME ..............................................  201   23.1       S/MIME Certificates .................................  201   23.2       S/MIME Key Exchange .................................  202   23.3       Securing MIME bodies ................................  205   23.4       SIP Header Privacy and Integrity using S/MIME:              Tunneling SIP .......................................  207   23.4.1     Integrity and Confidentiality Properties of SIP              Headers .............................................  207   23.4.1.1   Integrity ...........................................  207   23.4.1.2   Confidentiality .....................................  208   23.4.2     Tunneling Integrity and Authentication ..............  209   23.4.3     Tunneling Encryption ................................  211   24         Examples ............................................  213   24.1       Registration ........................................  213   24.2       Session Setup .......................................  214   25         Augmented BNF for the SIP Protocol ..................  219   25.1       Basic Rules .........................................  219   26         Security Considerations: Threat Model and Security              Usage Recommendations ...............................  232   26.1       Attacks and Threat Models ...........................  233   26.1.1     Registration Hijacking ..............................  233   26.1.2     Impersonating a Server ..............................  234   26.1.3     Tampering with Message Bodies .......................  235   26.1.4     Tearing Down Sessions ...............................  235Rosenberg, et. al.          Standards Track                     [Page 7]RFC 3261            SIP: Session Initiation Protocol           June 2002   26.1.5     Denial of Service and Amplification .................  236   26.2       Security Mechanisms .................................  237   26.2.1     Transport and Network Layer Security ................  238   26.2.2     SIPS URI Scheme .....................................  239   26.2.3     HTTP Authentication .................................  240   26.2.4     S/MIME ..............................................  240   26.3       Implementing Security Mechanisms ....................  241   26.3.1     Requirements for Implementers of SIP ................  241   26.3.2     Security Solutions ..................................  242   26.3.2.1   Registration ........................................  242   26.3.2.2   Interdomain Requests ................................  243   26.3.2.3   Peer-to-Peer Requests ...............................  245   26.3.2.4   DoS Protection ......................................  246   26.4       Limitations .........................................  247   26.4.1     HTTP Digest .........................................  247   26.4.2     S/MIME ..............................................  248   26.4.3     TLS .................................................  249   26.4.4     SIPS URIs ...........................................  249   26.5       Privacy .............................................  251   27         IANA Considerations .................................  252   27.1       Option Tags .........................................  252   27.2       Warn-Codes ..........................................  252   27.3       Header Field Names ..................................  253   27.4       Method and Response Codes ...........................  253   27.5       The "message/sip" MIME type.  .......................  254   27.6       New Content-Disposition Parameter Registrations .....  255   28         Changes From RFC 2543 ...............................  255   28.1       Major Functional Changes ............................  255   28.2       Minor Functional Changes ............................  260   29         Normative References ................................  261   30         Informative References ..............................  262   A          Table of Timer Values ...............................  265   Acknowledgments ................................................  266   Authors' Addresses .............................................  267   Full Copyright Statement .......................................  2691 Introduction   There are many applications of the Internet that require the creation   and management of a session, where a session is considered an   exchange of data between an association of participants.  The   implementation of these applications is complicated by the practices   of participants: users may move between endpoints, they may be   addressable by multiple names, and they may communicate in several   different media - sometimes simultaneously.  Numerous protocols have   been authored that carry various forms of real-time multimedia   session data such as voice, video, or text messages.  The Session   Initiation Protocol (SIP) works in concert with these protocols byRosenberg, et. al.          Standards Track                     [Page 8]RFC 3261            SIP: Session Initiation Protocol           June 2002   enabling Internet endpoints (called user agents) to discover one   another and to agree on a characterization of a session they would   like to share.  For locating prospective session participants, and   for other functions, SIP enables the creation of an infrastructure of   network hosts (called proxy servers) to which user agents can send   registrations, invitations to sessions, and other requests.  SIP is   an agile, general-purpose tool for creating, modifying, and   terminating sessions that works independently of underlying transport   protocols and without dependency on the type of session that is being   established.2 Overview of SIP Functionality   SIP is an application-layer control protocol that can establish,   modify, and terminate multimedia sessions (conferences) such as   Internet telephony calls.  SIP can also invite participants to   already existing sessions, such as multicast conferences.  Media can   be added to (and removed from) an existing session.  SIP   transparently supports name mapping and redirection services, which   supports personal mobility [27] - users can maintain a single   externally visible identifier regardless of their network location.   SIP supports five facets of establishing and terminating multimedia   communications:      User location: determination of the end system to be used for           communication;      User availability: determination of the willingness of the called           party to engage in communications;      User capabilities: determination of the media and media parameters           to be used;      Session setup: "ringing", establishment of session parameters at           both called and calling party;      Session management: including transfer and termination of           sessions, modifying session parameters, and invoking           services.   SIP is not a vertically integrated communications system.  SIP is   rather a component that can be used with other IETF protocols to   build a complete multimedia architecture.  Typically, these   architectures will include protocols such as the Real-time Transport   Protocol (RTP) (RFC 1889 [28]) for transporting real-time data and   providing QoS feedback, the Real-Time streaming protocol (RTSP) (RFC   2326 [29]) for controlling delivery of streaming media, the MediaRosenberg, et. al.          Standards Track                     [Page 9]RFC 3261            SIP: Session Initiation Protocol           June 2002   Gateway Control Protocol (MEGACO) (RFC 3015 [30]) for controlling   gateways to the Public Switched Telephone Network (PSTN), and the   Session Description Protocol (SDP) (RFC 2327 [1]) for describing   multimedia sessions.  Therefore, SIP should be used in conjunction   with other protocols in order to provide complete services to the   users.  However, the basic functionality and operation of SIP does   not depend on any of these protocols.   SIP does not provide services.  Rather, SIP provides primitives that   can be used to implement different services.  For example, SIP can   locate a user and deliver an opaque object to his current location.   If this primitive is used to deliver a session description written in   SDP, for instance, the endpoints can agree on the parameters of a   session.  If the same primitive is used to deliver a photo of the   caller as well as the session description, a "caller ID" service can   be easily implemented.  As this example shows, a single primitive is   typically used to provide several different services.   SIP does not offer conference control services such as floor control   or voting and does not prescribe how a conference is to be managed.   SIP can be used to initiate a session that uses some other conference   control protocol.  Since SIP messages and the sessions they establish   can pass through entirely different networks, SIP cannot, and does   not, provide any kind of network resource reservation capabilities.   The nature of the services provided make security particularly   important.  To that end, SIP provides a suite of security services,   which include denial-of-service prevention, authentication (both user   to user and proxy to user), integrity protection, and encryption and   privacy services.   SIP works with both IPv4 and IPv6.3 Terminology   In this document, the key words "MUST", "MUST NOT", "REQUIRED",   "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT   RECOMMENDED", "MAY", and "OPTIONAL" are to be interpreted as   described in BCP 14, RFC 2119 [2] and indicate requirement levels for   compliant SIP implementations.4 Overview of Operation   This section introduces the basic operations of SIP using simple   examples.  This section is tutorial in nature and does not contain   any normative statements.Rosenberg, et. al.          Standards Track                    [Page 10]RFC 3261            SIP: Session Initiation Protocol           June 2002   The first example shows the basic functions of SIP: location of an   end point, signal of a desire to communicate, negotiation of session   parameters to establish the session, and teardown of the session once   established.   Figure 1 shows a typical example of a SIP message exchange between   two users, Alice and Bob.  (Each message is labeled with the letter   "F" and a number for reference by the text.)  In this example, Alice   uses a SIP application on her PC (referred to as a softphone) to call   Bob on his SIP phone over the Internet.  Also shown are two SIP proxy   servers that act on behalf of Alice and Bob to facilitate the session   establishment.  This typical arrangement is often referred to as the   "SIP trapezoid" as shown by the geometric shape of the dotted lines   in Figure 1.   Alice "calls" Bob using his SIP identity, a type of Uniform Resource   Identifier (URI) called a SIP URI. SIP URIs are defined in Section   19.1.  It has a similar form to an email address, typically   containing a username and a host name.  In this case, it is   sip:bob@biloxi.com, where biloxi.com is the domain of Bob's SIP   service provider.  Alice has a SIP URI of sip:alice@atlanta.com.   Alice might have typed in Bob's URI or perhaps clicked on a hyperlink   or an entry in an address book.  SIP also provides a secure URI,   called a SIPS URI.  An example would be sips:bob@biloxi.com.  A call   made to a SIPS URI guarantees that secure, encrypted transport   (namely TLS) is used to carry all SIP messages from the caller to the   domain of the callee.  From there, the request is sent securely to   the callee, but with security mechanisms that depend on the policy of   the domain of the callee.   SIP is based on an HTTP-like request/response transaction model.   Each transaction consists of a request that invokes a particular   method, or function, on the server and at least one response.  In   this example, the transaction begins with Alice's softphone sending   an INVITE request addressed to Bob's SIP URI.  INVITE is an example   of a SIP method that specifies the action that the requestor (Alice)   wants the server (Bob) to take.  The INVITE request contains a number   of header fields.  Header fields are named attributes that provide   additional information about a message.  The ones present in an   INVITE include a unique identifier for the call, the destination   address, Alice's address, and information about the type of session   that Alice wishes to establish with Bob.  The INVITE (message F1 in   Figure 1) might look like this:Rosenberg, et. al.          Standards Track                    [Page 11]RFC 3261            SIP: Session Initiation Protocol           June 2002                     atlanta.com  . . . biloxi.com                 .      proxy              proxy     .               .                                       .       Alice's  . . . . . . . . . . . . . . . . . . . .  Bob's      softphone                                        SIP Phone         |                |                |                |         |    INVITE F1   |                |                |         |--------------->|    INVITE F2   |                |         |  100 Trying F3 |--------------->|    INVITE F4   |         |<---------------|  100 Trying F5 |--------------->|         |                |<-------------- | 180 Ringing F6 |         |                | 180 Ringing F7 |<---------------|         | 180 Ringing F8 |<---------------|     200 OK F9  |         |<---------------|    200 OK F10  |<---------------|         |    200 OK F11  |<---------------|                |         |<---------------|                |                |         |                       ACK F12                    |         |------------------------------------------------->|         |                   Media Session                  |         |<================================================>|         |                       BYE F13                    |         |<-------------------------------------------------|         |                     200 OK F14                   |         |------------------------------------------------->|         |                                                  |         Figure 1: SIP session setup example with SIP trapezoid      INVITE sip:bob@biloxi.com SIP/2.0      Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bK776asdhds      Max-Forwards: 70      To: Bob <sip:bob@biloxi.com>      From: Alice <sip:alice@atlanta.com>;tag=1928301774      Call-ID: a84b4c76e66710@pc33.atlanta.com      CSeq: 314159 INVITE      Contact: <sip:alice@pc33.atlanta.com>      Content-Type: application/sdp      Content-Length: 142      (Alice's SDP not shown)   The first line of the text-encoded message contains the method name   (INVITE).  The lines that follow are a list of header fields.  This   example contains a minimum required set.  The header fields are   briefly described below:Rosenberg, et. al.          Standards Track                    [Page 12]RFC 3261            SIP: Session Initiation Protocol           June 2002   Via contains the address (pc33.atlanta.com) at which Alice is   expecting to receive responses to this request.  It also contains a   branch parameter that identifies this transaction.   To contains a display name (Bob) and a SIP or SIPS URI   (sip:bob@biloxi.com) towards which the request was originally   directed.  Display names are described in RFC 2822 [3].   From also contains a display name (Alice) and a SIP or SIPS URI   (sip:alice@atlanta.com) that indicate the originator of the request.   This header field also has a tag parameter containing a random string   (1928301774) that was added to the URI by the softphone.  It is used   for identification purposes.   Call-ID contains a globally unique identifier for this call,   generated by the combination of a random string and the softphone's   host name or IP address.  The combination of the To tag, From tag,   and Call-ID completely defines a peer-to-peer SIP relationship   between Alice and Bob and is referred to as a dialog.   CSeq or Command Sequence contains an integer and a method name.  The   CSeq number is incremented for each new request within a dialog and   is a traditional sequence number.   Contact contains a SIP or SIPS URI that represents a direct route to   contact Alice, usually composed of a username at a fully qualified   domain name (FQDN).  While an FQDN is preferred, many end systems do   not have registered domain names, so IP addresses are permitted.   While the Via header field tells other elements where to send the   response, the Contact header field tells other elements where to send   future requests.   Max-Forwards serves to limit the number of hops a request can make on   the way to its destination.  It consists of an integer that is   decremented by one at each hop.   Content-Type contains a description of the message body (not shown).   Content-Length contains an octet (byte) count of the message body.   The complete set of SIP header fields is defined in Section 20.   The details of the session, such as the type of media, codec, or   sampling rate, are not described using SIP.  Rather, the body of a   SIP message contains a description of the session, encoded in some   other protocol format.  One such format is the Session Description   Protocol (SDP) (RFC 2327 [1]).  This SDP message (not shown in theRosenberg, et. al.          Standards Track                    [Page 13]RFC 3261            SIP: Session Initiation Protocol           June 2002   example) is carried by the SIP message in a way that is analogous to   a document attachment being carried by an email message, or a web   page being carried in an HTTP message.   Since the softphone does not know the location of Bob or the SIP   server in the biloxi.com domain, the softphone sends the INVITE to   the SIP server that serves Alice's domain, atlanta.com.  The address   of the atlanta.com SIP server could have been configured in Alice's   softphone, or it could have been discovered by DHCP, for example.   The atlanta.com SIP server is a type of SIP server known as a proxy   server.  A proxy server receives SIP requests and forwards them on   behalf of the requestor.  In this example, the proxy server receives   the INVITE request and sends a 100 (Trying) response back to Alice's   softphone.  The 100 (Trying) response indicates that the INVITE has   been received and that the proxy is working on her behalf to route   the INVITE to the destination.  Responses in SIP use a three-digit   code followed by a descriptive phrase.  This response contains the   same To, From, Call-ID, CSeq and branch parameter in the Via as the   INVITE, which allows Alice's softphone to correlate this response to   the sent INVITE.  The atlanta.com proxy server locates the proxy   server at biloxi.com, possibly by performing a particular type of DNS   (Domain Name Service) lookup to find the SIP server that serves the   biloxi.com domain.  This is described in [4].  As a result, it   obtains the IP address of the biloxi.com proxy server and forwards,   or proxies, the INVITE request there.  Before forwarding the request,   the atlanta.com proxy server adds an additional Via header field   value that contains its own address (the INVITE already contains   Alice's address in the first Via).  The biloxi.com proxy server   receives the INVITE and responds with a 100 (Trying) response back to   the atlanta.com proxy server to indicate that it has received the   INVITE and is processing the request.  The proxy server consults a   database, generically called a location service, that contains the   current IP address of Bob.  (We shall see in the next section how   this database can be populated.)  The biloxi.com proxy server adds   another Via header field value with its own address to the INVITE and   proxies it to Bob's SIP phone.   Bob's SIP phone receives the INVITE and alerts Bob to the incoming   call from Alice so that Bob can decide whether to answer the call,   that is, Bob's phone rings.  Bob's SIP phone indicates this in a 180   (Ringing) response, which is routed back through the two proxies in   the reverse direction.  Each proxy uses the Via header field to   determine where to send the response and removes its own address from   the top.  As a result, although DNS and location service lookups were   required to route the initial INVITE, the 180 (Ringing) response can   be returned to the caller without lookups or without state beingRosenberg, et. al.          Standards Track                    [Page 14]RFC 3261            SIP: Session Initiation Protocol           June 2002   maintained in the proxies.  This also has the desirable property that   each proxy that sees the INVITE will also see all responses to the   INVITE.   When Alice's softphone receives the 180 (Ringing) response, it passes   this information to Alice, perhaps using an audio ringback tone or by   displaying a message on Alice's screen.   In this example, Bob decides to answer the call.  When he picks up   the handset, his SIP phone sends a 200 (OK) response to indicate that   the call has been answered.  The 200 (OK) contains a message body   with the SDP media description of the type of session that Bob is   willing to establish with Alice.  As a result, there is a two-phase   exchange of SDP messages: Alice sent one to Bob, and Bob sent one   back to Alice.  This two-phase exchange provides basic negotiation   capabilities and is based on a simple offer/answer model of SDP   exchange.  If Bob did not wish to answer the call or was busy on   another call, an error response would have been sent instead of the   200 (OK), which would have resulted in no media session being   established.  The complete list of SIP response codes is in Section   21.  The 200 (OK) (message F9 in Figure 1) might look like this as   Bob sends it out:      SIP/2.0 200 OK      Via: SIP/2.0/UDP server10.biloxi.com         ;branch=z9hG4bKnashds8;received=192.0.2.3      Via: SIP/2.0/UDP bigbox3.site3.atlanta.com         ;branch=z9hG4bK77ef4c2312983.1;received=192.0.2.2      Via: SIP/2.0/UDP pc33.atlanta.com         ;branch=z9hG4bK776asdhds ;received=192.0.2.1      To: Bob <sip:bob@biloxi.com>;tag=a6c85cf      From: Alice <sip:alice@atlanta.com>;tag=1928301774      Call-ID: a84b4c76e66710@pc33.atlanta.com      CSeq: 314159 INVITE      Contact: <sip:bob@192.0.2.4>      Content-Type: application/sdp      Content-Length: 131      (Bob's SDP not shown)   The first line of the response contains the response code (200) and   the reason phrase (OK).  The remaining lines contain header fields.   The Via, To, From, Call-ID, and CSeq header fields are copied from   the INVITE request.  (There are three Via header field values - one   added by Alice's SIP phone, one added by the atlanta.com proxy, and   one added by the biloxi.com proxy.)  Bob's SIP phone has added a tag   parameter to the To header field.  This tag will be incorporated by   both endpoints into the dialog and will be included in all futureRosenberg, et. al.          Standards Track                    [Page 15]RFC 3261            SIP: Session Initiation Protocol           June 2002   requests and responses in this call.  The Contact header field   contains a URI at which Bob can be directly reached at his SIP phone.   The Content-Type and Content-Length refer to the message body (not   shown) that contains Bob's SDP media information.   In addition to DNS and location service lookups shown in this   example, proxy servers can make flexible "routing decisions" to   decide where to send a request.  For example, if Bob's SIP phone   returned a 486 (Busy Here) response, the biloxi.com proxy server   could proxy the INVITE to Bob's voicemail server.  A proxy server can   also send an INVITE to a number of locations at the same time.  This   type of parallel search is known as forking.   In this case, the 200 (OK) is routed back through the two proxies and   is received by Alice's softphone, which then stops the ringback tone   and indicates that the call has been answered.  Finally, Alice's   softphone sends an acknowledgement message, ACK, to Bob's SIP phone   to confirm the reception of the final response (200 (OK)).  In this   example, the ACK is sent directly from Alice's softphone to Bob's SIP   phone, bypassing the two proxies.  This occurs because the endpoints   have learned each other's address from the Contact header fields   through the INVITE/200 (OK) exchange, which was not known when the   initial INVITE was sent.  The lookups performed by the two proxies   are no longer needed, so the proxies drop out of the call flow.  This   completes the INVITE/200/ACK three-way handshake used to establish   SIP sessions.  Full details on session setup are in Section 13.   Alice and Bob's media session has now begun, and they send media   packets using the format to which they agreed in the exchange of SDP.   In general, the end-to-end media packets take a different path from   the SIP signaling messages.   During the session, either Alice or Bob may decide to change the   characteristics of the media session.  This is accomplished by   sending a re-INVITE containing a new media description.  This re-   INVITE references the existing dialog so that the other party knows   that it is to modify an existing session instead of establishing a   new session.  The other party sends a 200 (OK) to accept the change.   The requestor responds to the 200 (OK) with an ACK.  If the other   party does not accept the change, he sends an error response such as   488 (Not Acceptable Here), which also receives an ACK.  However, the   failure of the re-INVITE does not cause the existing call to fail -   the session continues using the previously negotiated   characteristics.  Full details on session modification are in Section   14.Rosenberg, et. al.          Standards Track                    [Page 16]RFC 3261            SIP: Session Initiation Protocol           June 2002   At the end of the call, Bob disconnects (hangs up) first and   generates a BYE message.  This BYE is routed directly to Alice's   softphone, again bypassing the proxies.  Alice confirms receipt of   the BYE with a 200 (OK) response, which terminates the session and   the BYE transaction.  No ACK is sent - an ACK is only sent in   response to a response to an INVITE request.  The reasons for this   special handling for INVITE will be discussed later, but relate to   the reliability mechanisms in SIP, the length of time it can take for   a ringing phone to be answered, and forking.  For this reason,   request handling in SIP is often classified as either INVITE or non-   INVITE, referring to all other methods besides INVITE.  Full details   on session termination are in Section 15.   Section 24.2 describes the messages shown in Figure 1 in full.   In some cases, it may be useful for proxies in the SIP signaling path   to see all the messaging between the endpoints for the duration of   the session.  For example, if the biloxi.com proxy server wished to   remain in the SIP messaging path beyond the initial INVITE, it would   add to the INVITE a required routing header field known as Record-   Route that contained a URI resolving to the hostname or IP address of   the proxy.  This information would be received by both Bob's SIP   phone and (due to the Record-Route header field being passed back in   the 200 (OK)) Alice's softphone and stored for the duration of the   dialog.  The biloxi.com proxy server would then receive and proxy the   ACK, BYE, and 200 (OK) to the BYE.  Each proxy can independently   decide to receive subsequent messages, and those messages will pass   through all proxies that elect to receive it.  This capability is   frequently used for proxies that are providing mid-call features.   Registration is another common operation in SIP.  Registration is one   way that the biloxi.com server can learn the current location of Bob.   Upon initialization, and at periodic intervals, Bob's SIP phone sends   REGISTER messages to a server in the biloxi.com domain known as a SIP   registrar.  The REGISTER messages associate Bob's SIP or SIPS URI   (sip:bob@biloxi.com) with the machine into which he is currently   logged (conveyed as a SIP or SIPS URI in the Contact header field).   The registrar writes this association, also called a binding, to a   database, called the location service, where it can be used by the   proxy in the biloxi.com domain.  Often, a registrar server for a   domain is co-located with the proxy for that domain.  It is an   important concept that the distinction between types of SIP servers   is logical, not physical.   Bob is not limited to registering from a single device.  For example,   both his SIP phone at home and the one in the office could send   registrations.  This information is stored together in the locationRosenberg, et. al.          Standards Track                    [Page 17]RFC 3261            SIP: Session Initiation Protocol           June 2002   service and allows a proxy to perform various types of searches to   locate Bob.  Similarly, more than one user can be registered on a   single device at the same time.   The location service is just an abstract concept.  It generally   contains information that allows a proxy to input a URI and receive a   set of zero or more URIs that tell the proxy where to send the   request.  Registrations are one way to create this information, but   not the only way.  Arbitrary mapping functions can be configured at   the discretion of the administrator.   Finally, it is important to note that in SIP, registration is used   for routing incoming SIP requests and has no role in authorizing   outgoing requests.  Authorization and authentication are handled in   SIP either on a request-by-request basis with a challenge/response   mechanism, or by using a lower layer scheme as discussed in Section   26.   The complete set of SIP message details for this registration example   is in Section 24.1.   Additional operations in SIP, such as querying for the capabilities   of a SIP server or client using OPTIONS, or canceling a pending   request using CANCEL, will be introduced in later sections.5 Structure of the Protocol   SIP is structured as a layered protocol, which means that its   behavior is described in terms of a set of fairly independent   processing stages with only a loose coupling between each stage.  The   protocol behavior is described as layers for the purpose of   presentation, allowing the description of functions common across   elements in a single section.  It does not dictate an implementation   in any way.  When we say that an element "contains" a layer, we mean   it is compliant to the set of rules defined by that layer.   Not every element specified by the protocol contains every layer.   Furthermore, the elements specified by SIP are logical elements, not   physical ones.  A physical realization can choose to act as different   logical elements, perhaps even on a transaction-by-transaction basis.   The lowest layer of SIP is its syntax and encoding.  Its encoding is   specified using an augmented Backus-Naur Form grammar (BNF).  The   complete BNF is specified in Section 25; an overview of a SIP   message's structure can be found in Section 7.Rosenberg, et. al.          Standards Track                    [Page 18]RFC 3261            SIP: Session Initiation Protocol           June 2002   The second layer is the transport layer.  It defines how a client   sends requests and receives responses and how a server receives   requests and sends responses over the network.  All SIP elements   contain a transport layer.  The transport layer is described in   Section 18.   The third layer is the transaction layer.  Transactions are a   fundamental component of SIP.  A transaction is a request sent by a   client transaction (using the transport layer) to a server   transaction, along with all responses to that request sent from the   server transaction back to the client.  The transaction layer handles   application-layer retransmissions, matching of responses to requests,   and application-layer timeouts.  Any task that a user agent client   (UAC) accomplishes takes place using a series of transactions.   Discussion of transactions can be found in Section 17.  User agents   contain a transaction layer, as do stateful proxies.  Stateless   proxies do not contain a transaction layer.  The transaction layer   has a client component (referred to as a client transaction) and a   server component (referred to as a server transaction), each of which   are represented by a finite state machine that is constructed to   process a particular request.   The layer above the transaction layer is called the transaction user   (TU).  Each of the SIP entities, except the stateless proxy, is a   transaction user.  When a TU wishes to send a request, it creates a   client transaction instance and passes it the request along with the   destination IP address, port, and transport to which to send the   request.  A TU that creates a client transaction can also cancel it.   When a client cancels a transaction, it requests that the server stop   further processing, revert to the state that existed before the   transaction was initiated, and generate a specific error response to   that transaction.  This is done with a CANCEL request, which   constitutes its own transaction, but references the transaction to be   cancelled (Section 9).   The SIP elements, that is, user agent clients and servers, stateless   and stateful proxies and registrars, contain a core that   distinguishes them from each other.  Cores, except for the stateless   proxy, are transaction users.  While the behavior of the UAC and UAS   cores depends on the method, there are some common rules for all   methods (Section 8).  For a UAC, these rules govern the construction   of a request; for a UAS, they govern the processing of a request and   generating a response.  Since registrations play an important role in   SIP, a UAS that handles a REGISTER is given the special name   registrar.  Section 10 describes UAC and UAS core behavior for the   REGISTER method.  Section 11 describes UAC and UAS core behavior for   the OPTIONS method, used for determining the capabilities of a UA.Rosenberg, et. al.          Standards Track                    [Page 19]RFC 3261            SIP: Session Initiation Protocol           June 2002   Certain other requests are sent within a dialog.  A dialog is a   peer-to-peer SIP relationship between two user agents that persists   for some time.  The dialog facilitates sequencing of messages and   proper routing of requests between the user agents.  The INVITE   method is the only way defined in this specification to establish a   dialog.  When a UAC sends a request that is within the context of a   dialog, it follows the common UAC rules as discussed in Section 8 but   also the rules for mid-dialog requests.  Section 12 discusses dialogs   and presents the procedures for their construction and maintenance,   in addition to construction of requests within a dialog.   The most important method in SIP is the INVITE method, which is used   to establish a session between participants.  A session is a   collection of participants, and streams of media between them, for   the purposes of communication.  Section 13 discusses how sessions are   initiated, resulting in one or more SIP dialogs.  Section 14   discusses how characteristics of that session are modified through   the use of an INVITE request within a dialog.  Finally, section 15   discusses how a session is terminated.   The procedures of Sections 8, 10, 11, 12, 13, 14, and 15 deal   entirely with the UA core (Section 9 describes cancellation, which   applies to both UA core and proxy core).  Section 16 discusses the   proxy element, which facilitates routing of messages between user   agents.6 Definitions   The following terms have special significance for SIP.      Address-of-Record: An address-of-record (AOR) is a SIP or SIPS URI         that points to a domain with a location service that can map         the URI to another URI where the user might be available.         Typically, the location service is populated through         registrations.  An AOR is frequently thought of as the "public         address" of the user.      Back-to-Back User Agent: A back-to-back user agent (B2BUA) is a         logical entity that receives a request and processes it as a         user agent server (UAS).  In order to determine how the request         should be answered, it acts as a user agent client (UAC) and         generates requests.  Unlike a proxy server, it maintains dialog         state and must participate in all requests sent on the dialogs         it has established.  Since it is a concatenation of a UAC and         UAS, no explicit definitions are needed for its behavior.Rosenberg, et. al.          Standards Track                    [Page 20]RFC 3261            SIP: Session Initiation Protocol           June 2002      Call: A call is an informal term that refers to some communication         between peers, generally set up for the purposes of a         multimedia conversation.      Call Leg: Another name for a dialog [31]; no longer used in this         specification.      Call Stateful: A proxy is call stateful if it retains state for a         dialog from the initiating INVITE to the terminating BYE         request.  A call stateful proxy is always transaction stateful,         but the converse is not necessarily true.      Client: A client is any network element that sends SIP requests         and receives SIP responses.  Clients may or may not interact         directly with a human user.  User agent clients and proxies are         clients.      Conference: A multimedia session (see below) that contains         multiple participants.      Core: Core designates the functions specific to a particular type         of SIP entity, i.e., specific to either a stateful or stateless         proxy, a user agent or registrar.  All cores, except those for         the stateless proxy, are transaction users.      Dialog: A dialog is a peer-to-peer SIP relationship between two         UAs that persists for some time.  A dialog is established by         SIP messages, such as a 2xx response to an INVITE request.  A         dialog is identified by a call identifier, local tag, and a         remote tag.  A dialog was formerly known as a call leg in RFC         2543.      Downstream: A direction of message forwarding within a transaction         that refers to the direction that requests flow from the user         agent client to user agent server.      Final Response: A response that terminates a SIP transaction, as         opposed to a provisional response that does not.  All 2xx, 3xx,         4xx, 5xx and 6xx responses are final.      Header: A header is a component of a SIP message that conveys         information about the message.  It is structured as a sequence         of header fields.      Header Field: A header field is a component of the SIP message         header.  A header field can appear as one or more header field         rows. Header field rows consist of a header field name and zero         or more header field values. Multiple header field values on aRosenberg, et. al.          Standards Track                    [Page 21]RFC 3261            SIP: Session Initiation Protocol           June 2002         given header field row are separated by commas. Some header         fields can only have a single header field value, and as a         result, always appear as a single header field row.      Header Field Value: A header field value is a single value; a         header field consists of zero or more header field values.      Home Domain: The domain providing service to a SIP user.         Typically, this is the domain present in the URI in the         address-of-record of a registration.      Informational Response: Same as a provisional response.      Initiator, Calling Party, Caller: The party initiating a session         (and dialog) with an INVITE request.  A caller retains this         role from the time it sends the initial INVITE that established         a dialog until the termination of that dialog.      Invitation: An INVITE request.      Invitee, Invited User, Called Party, Callee: The party that         r